diff options
Diffstat (limited to 'third_party/libwebrtc/webrtc/audio/null_audio_poller.cc')
-rw-r--r-- | third_party/libwebrtc/webrtc/audio/null_audio_poller.cc | 66 |
1 files changed, 66 insertions, 0 deletions
diff --git a/third_party/libwebrtc/webrtc/audio/null_audio_poller.cc b/third_party/libwebrtc/webrtc/audio/null_audio_poller.cc new file mode 100644 index 0000000000..c22b3d8791 --- /dev/null +++ b/third_party/libwebrtc/webrtc/audio/null_audio_poller.cc @@ -0,0 +1,66 @@ +/* + * Copyright (c) 2017 The WebRTC project authors. All Rights Reserved. + * + * Use of this source code is governed by a BSD-style license + * that can be found in the LICENSE file in the root of the source + * tree. An additional intellectual property rights grant can be found + * in the file PATENTS. All contributing project authors may + * be found in the AUTHORS file in the root of the source tree. + */ + +#include "audio/null_audio_poller.h" +#include "rtc_base/logging.h" +#include "rtc_base/thread.h" + +namespace webrtc { +namespace internal { + +namespace { + +constexpr int64_t kPollDelayMs = 10; // WebRTC uses 10ms by default + +constexpr size_t kNumChannels = 1; +constexpr uint32_t kSamplesPerSecond = 48000; // 48kHz +constexpr size_t kNumSamples = kSamplesPerSecond / 100; // 10ms of samples + +} // namespace + +NullAudioPoller::NullAudioPoller(AudioTransport* audio_transport) + : audio_transport_(audio_transport), + reschedule_at_(rtc::TimeMillis() + kPollDelayMs) { + RTC_DCHECK(audio_transport); + OnMessage(nullptr); // Start the poll loop. +} + +NullAudioPoller::~NullAudioPoller() { + RTC_DCHECK(thread_checker_.CalledOnValidThread()); + rtc::Thread::Current()->Clear(this); +} + +void NullAudioPoller::OnMessage(rtc::Message* msg) { + RTC_DCHECK(thread_checker_.CalledOnValidThread()); + + // Buffer to hold the audio samples. + int16_t buffer[kNumSamples * kNumChannels]; + // Output variables from |NeedMorePlayData|. + size_t n_samples; + int64_t elapsed_time_ms; + int64_t ntp_time_ms; + audio_transport_->NeedMorePlayData(kNumSamples, sizeof(int16_t), kNumChannels, + kSamplesPerSecond, buffer, n_samples, + &elapsed_time_ms, &ntp_time_ms); + + // Reschedule the next poll iteration. If, for some reason, the given + // reschedule time has already passed, reschedule as soon as possible. + int64_t now = rtc::TimeMillis(); + if (reschedule_at_ < now) { + reschedule_at_ = now; + } + rtc::Thread::Current()->PostAt(RTC_FROM_HERE, reschedule_at_, this, 0); + + // Loop after next will be kPollDelayMs later. + reschedule_at_ += kPollDelayMs; +} + +} // namespace internal +} // namespace webrtc |